A Microfiche Appendix of the code is attached and comprises one (1) sheet having a total of forty five (45) frames.
The following applications are hereby incorporated by reference in their entirety and made part of the present application:
1. U.S. Provisional Application Serial No. 60/097,569, entitled xe2x80x9cAdaptive Rate Speech Codec,xe2x80x9d filed Aug. 24, 1998;
2. U.S. patent application Ser. No. 09/154,675, entitled xe2x80x9cSpeech Encoder Using Continuous Warping In Long Term Preprocessing,xe2x80x9d filed Sep. 18, 1998;
3. U.S. patent application Ser. No. 09/156,814, entitled xe2x80x9cSelectable Mode Vocoder System,xe2x80x9d filed Sep. 18, 1998;
4. U.S. patent application Ser. No. 09/156,649, entitled xe2x80x9cComb Codebook Structure,xe2x80x9d filed Sep. 18, 1998;
5. U.S. patent application Ser. No. 09/156,648, entitled xe2x80x9cLow Complexity Random Codebook Structure,xe2x80x9d filed Sep. 18, 1998;
6. U.S. patent application Ser. No. 09/156,650, entitled xe2x80x9cSpeech Encoder Using Gain Normalization that Combines Open and Closed Loop Gains,xe2x80x9d filed Sep. 18, 1998;
7. U.S. patent application Ser. No. 09/156,832, entitled xe2x80x9cSpeech Encoder Using Voice Activity Detection in Coding Noise,xe2x80x9d filed Sep. 18, 1998;
8. U.S. patent application Ser. No. 09/154,660, entitled xe2x80x9cSpeech Encoder Adaptively Applying Pitch Processing with Continuous Warping,xe2x80x9d filed Sep. 18, 1998;
9. U.S. patent application Ser. No. 09/154,654, entitled xe2x80x9cPitch Determination Using Speech Classification and Prior Pitch Estimation,xe2x80x9d filed Sep. 18, 1998;
10. U.S. patent application Ser. No. 09/154,657, entitled xe2x80x9cSpeech Encoder Using A Classifier For Smoothing Noise Coding,xe2x80x9d filed Sep. 18, 1998;
11. U.S. patent application Ser. No. 09/154,663, entitled xe2x80x9cAdaptive Gain Reduction To Produce Fixed Codebook Target Signal,xe2x80x9d filed Sep. 18, 1998;
12. U.S. patent application Ser. No. 09/154,662, entitled xe2x80x9cSpeech Classification and Parameter Weighting Used in Codebook Search,xe2x80x9d filed Sep. 18, 1998;
13. U.S. patent application Ser. No. 09/154,653, entitled xe2x80x9cSynchronized Encoder-Decoder Frame Concealment Using Speech Coding Parameters,xe2x80x9d filed Sep. 18, 1998;
14. U.S. patent application Ser. No. 09/157,083, entitled xe2x80x9cRobust Fast Search For Two-Dimensional Gain Vector Quantizer,xe2x80x9d filed Sep. 18, 1998;
15. U.S. patent application Ser. No. 09/156,416, entitled xe2x80x9cMethod and Apparatus for Detecting Voice Activity in a Speech Signal,xe2x80x9d filed Sep. 18, 1998.
1. Technical Field
The present invention relates generally to speech encoding and decoding in voice communication systems; and, more particularly, it relates to various techniques used with code-excited linear prediction coding to obtain high quality speech reproduction through a limited bit rate communication channel.
2. Related Art
Signal modeling and parameter estimation play significant roles in communicating voice information with limited bandwidth constraints. To model basic speech sounds, speech signals are sampled as a discrete waveform to be digitally processed. In one type of signal coding technique called LPC (linear predictive coding), the signal value at any particular time index is modeled as a linear function of previous values. A subsequent signal is thus linearly predictable according to an earlier value. As a result, efficient signal representations can be determined by estimating and applying certain prediction parameters to represent the signal.
Applying LPC techniques, a conventional source encoder operates on speech signals to extract modeling and parameter information for communication to a conventional source decoder via a communication channel. Once received, the decoder attempts to reconstruct a counterpart signal for playback that sounds to a human ear like the original speech.
A certain amount of communication channel bandwidth is required to communicate the modeling and parameter information to the decoder. In embodiments, for example where the channel bandwidth is shared and real-time reconstruction is necessary, a reduction in the required bandwidth proves beneficial. However, using conventional modeling techniques, the quality requirements in the reproduced speech limit the reduction of such bandwidth below certain levels.
In conventional code-excited linear predictive coding, waveform matching in the high frequency region proves more difficult than matching in the low frequency region. Thus, the energy of the high frequency region of a synthesized speech signal drops more than in the low frequency region, especially for low bit rate coding. Moreover, the amount of high frequency energy drop is not consistent. As a result, with conventional, lower bit rate speech codecs, reproduced speech signals exhibit poor (dull) sound quality.
Further limitations and disadvantages of conventional systems will become apparent to one of skill in the art after reviewing the remainder of the present application with reference to the drawings.
Various aspects of the present invention can be found in a speech system using an analysis by synthesis approach on a speech signal. The speech system comprises at least one codebook, containing at least one code vector, and processing circuitry. Using the at least one codebook, the processing circuitry generates a synthesized residual signal. The processing circuitry applies adaptive tilt compensation to the synthesized residual signal. The processing circuitry may also comprise both an encoder processing circuit that generates the synthesized residual signal, and a decoder processing circuit that applies the adaptive tilt compensation.
In other variations, the synthesized residual signal is a weighted synthesized residual signal. The adaptive tilt compensation may involve identification of a filter coefficient for use in a compensating filter, e.g., a first order filter. Such identification can be carried out by applying a window to the synthesized residual.
Further aspects of the present invention may be found in a speech system that also uses an analysis by synthesis approach on a speech signal. Therein, in addition to a codebook, a first processing circuit and second processing circuit can be found. The first processing circuit generates both a residual signal and, using the codebook, a synthesized residual signal. Both of these signals may be weighted. The residual signal has a first spectral envelope, while the synthesized residual has a second spectral envelope that exhibits variations from the first. The second processing circuit adaptively attempts to minimize such variations. In at least some embodiments, the attempt is made without having access to the residual signal. Of course, at least most of the aforementioned variations are equally applicable to the present speech system.
Other aspects, advantages and novel features of the present invention will become apparent from the following detailed description of the invention when considered in conjunction with the accompanying drawings.